[asterisk-commits] oej: branch oej/adb-appleraision-1.8-mark-2 r415057 - in /team/oej/adb-appler...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Tue Jun 3 11:07:06 CDT 2014


Author: oej
Date: Tue Jun  3 11:06:55 2014
New Revision: 415057

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=415057
Log:
Yay subversion

Quoting Matt Jordan

Modified:
    team/oej/adb-appleraision-1.8-mark-2/   (props changed)
    team/oej/adb-appleraision-1.8-mark-2/build_tools/make_version
    team/oej/adb-appleraision-1.8-mark-2/configs/agents.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/asterisk.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/chan_dahdi.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/dsp.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/extconfig.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/h323.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/iax.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/indications.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/manager.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/queues.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/res_curl.conf.sample   (props changed)
    team/oej/adb-appleraision-1.8-mark-2/configs/res_fax.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/res_ldap.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/res_odbc.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/sip.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/sip_notify.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/sla.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/configs/voicemail.conf.sample
    team/oej/adb-appleraision-1.8-mark-2/funcs/func_odbc.c
    team/oej/adb-appleraision-1.8-mark-2/main/adsi.c
    team/oej/adb-appleraision-1.8-mark-2/main/aoc.c
    team/oej/adb-appleraision-1.8-mark-2/main/asterisk.exports.in
    team/oej/adb-appleraision-1.8-mark-2/main/astfd.c
    team/oej/adb-appleraision-1.8-mark-2/main/astobj2.c
    team/oej/adb-appleraision-1.8-mark-2/main/audiohook.c
    team/oej/adb-appleraision-1.8-mark-2/main/callerid.c
    team/oej/adb-appleraision-1.8-mark-2/main/ccss.c
    team/oej/adb-appleraision-1.8-mark-2/main/cel.c
    team/oej/adb-appleraision-1.8-mark-2/main/cli.c
    team/oej/adb-appleraision-1.8-mark-2/main/devicestate.c
    team/oej/adb-appleraision-1.8-mark-2/main/dnsmgr.c
    team/oej/adb-appleraision-1.8-mark-2/main/editline/readline.c
    team/oej/adb-appleraision-1.8-mark-2/main/editline/term.c
    team/oej/adb-appleraision-1.8-mark-2/main/enum.c
    team/oej/adb-appleraision-1.8-mark-2/main/file.c
    team/oej/adb-appleraision-1.8-mark-2/main/frame.c
    team/oej/adb-appleraision-1.8-mark-2/main/heap.c
    team/oej/adb-appleraision-1.8-mark-2/main/indications.c
    team/oej/adb-appleraision-1.8-mark-2/main/io.c
    team/oej/adb-appleraision-1.8-mark-2/main/lock.c
    team/oej/adb-appleraision-1.8-mark-2/main/logger.c
    team/oej/adb-appleraision-1.8-mark-2/main/netsock.c
    team/oej/adb-appleraision-1.8-mark-2/main/pbx.c
    team/oej/adb-appleraision-1.8-mark-2/main/rtp_engine.c
    team/oej/adb-appleraision-1.8-mark-2/main/say.c
    team/oej/adb-appleraision-1.8-mark-2/main/sched.c
    team/oej/adb-appleraision-1.8-mark-2/main/security_events.c
    team/oej/adb-appleraision-1.8-mark-2/main/strcompat.c
    team/oej/adb-appleraision-1.8-mark-2/main/stun.c
    team/oej/adb-appleraision-1.8-mark-2/main/taskprocessor.c
    team/oej/adb-appleraision-1.8-mark-2/main/tcptls.c
    team/oej/adb-appleraision-1.8-mark-2/main/threadstorage.c
    team/oej/adb-appleraision-1.8-mark-2/main/udptl.c
    team/oej/adb-appleraision-1.8-mark-2/main/xmldoc.c
    team/oej/adb-appleraision-1.8-mark-2/makeopts.in
    team/oej/adb-appleraision-1.8-mark-2/res/ael/pval.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_agi.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_calendar.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_calendar_ews.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_config_ldap.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_config_odbc.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_config_sqlite.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_mutestream.c   (contents, props changed)
    team/oej/adb-appleraision-1.8-mark-2/res/res_odbc.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_odbc.exports.in
    team/oej/adb-appleraision-1.8-mark-2/res/res_rtp_multicast.c   (contents, props changed)
    team/oej/adb-appleraision-1.8-mark-2/res/res_srtp.c
    team/oej/adb-appleraision-1.8-mark-2/res/res_timing_pthread.c

Propchange: team/oej/adb-appleraision-1.8-mark-2/
------------------------------------------------------------------------------
    automerge = Is-there-life-off-net?

Propchange: team/oej/adb-appleraision-1.8-mark-2/
            ('svnmerge-integrated' removed)

Modified: team/oej/adb-appleraision-1.8-mark-2/build_tools/make_version
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/build_tools/make_version?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/build_tools/make_version (original)
+++ team/oej/adb-appleraision-1.8-mark-2/build_tools/make_version Tue Jun  3 11:06:55 2014
@@ -37,9 +37,6 @@
 
         if [ ${BRANCH} != 0 ] ; then
             RESULT="${RESULT}-${PART}"
-            if [ ${FEATURE} != 0 ] ; then
-                RESULT="${RESULT}-${FEATURE_NAME}"
-            fi
             if [ ${FEATURE} != 0 ] ; then
                 RESULT="${RESULT}-${FEATURE_NAME}"
             fi

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/agents.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/agents.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/agents.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/agents.conf.sample Tue Jun  3 11:06:55 2014
@@ -3,9 +3,6 @@
 ;
 
 [general]
-; Enable or disable a single extension from logging in as multiple agents.
-; The default value is "yes".
-;multiplelogin=yes
 
 [agents]
 ;
@@ -30,7 +27,7 @@
 ;autologoffunavail=yes
 ;
 ; Define ackcall to require a DTMF acknowledgement when
-; an agent logs in using AgentLogin.  Default is "no".
+; a logged-in agent receives a call.  Default is "no".
 ; Use the acceptdtmf option to configure what DTMF key
 ; press should be used to acknowledge the call. The
 ; default is '#'.
@@ -56,11 +53,6 @@
 ; musiconhold => music_class
 ;
 ;musiconhold => default
-;
-; Define the default good bye sound file for agents
-; default to vm-goodbye
-;
-;goodbye => goodbye_file
 ;
 ; Define updatecdr. This is whether or not to change the source
 ; channel in the CDR record for this call to agent/agent_id so

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/asterisk.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/asterisk.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/asterisk.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/asterisk.conf.sample Tue Jun  3 11:06:55 2014
@@ -27,7 +27,6 @@
 ;dontwarn = yes			; Disable some warnings.
 ;dumpcore = yes			; Dump core on crash (same as -g at startup).
 ;languageprefix = yes		; Use the new sound prefix path syntax.
-;internal_timing = yes
 ;systemname = my_system_name	; Prefix uniqueid with a system name for
 				; Global uniqueness issues.
 ;autosystemname = yes		; Automatically set systemname to hostname,
@@ -73,6 +72,12 @@
 ;lockconfdir = no		; Protect the directory containing the
 				; configuration files (/etc/asterisk) with a
 				; lock.
+;live_dangerously = no		; Enable the execution of 'dangerous' dialplan
+				; functions from external sources (AMI,
+				; etc.) These functions (such as SHELL) are
+				; considered dangerous because they can allow
+				; privilege escalation.
+				; Default yes, for backward compatability.
 
 ; Changing the following lines may compromise your security.
 ;[files]

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/chan_dahdi.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/chan_dahdi.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/chan_dahdi.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/chan_dahdi.conf.sample Tue Jun  3 11:06:55 2014
@@ -168,6 +168,18 @@
 ; B channels; defaults to 'never'.
 ;
 ;resetinterval = 3600
+;
+; Assume inband audio may be present when a PROCEEDING message is received.
+; Q.931 Section 5.1.2 says the network cannot assume that the CPE side has
+; attached to the B channel at this time without explicitly sending the
+; progress indicator ie informing the CPE side to attach to the B channel
+; for audio.  However, some non-compliant ISDN switches send a PROCEEDING
+; without the progress indicator ie indicating inband audio is available and
+; assume that the CPE device has connected the media path for listening to
+; ringback and other messages.
+; Default yes in current release branches for backward compatibility.
+;
+;inband_on_proceeding=yes
 ;
 ; Overlap dialing mode (sending overlap digits)
 ; Cannot be changed on a reload.
@@ -491,7 +503,8 @@
 ; easily be re-attaching to a prior incoming call that was not yet hung up).
 ; This option changes the hangup to wait for a dialtone on the line, before
 ; marking the line as once again available for use with outgoing calls.
-;waitfordialtone=yes
+; Specified in milliseconds, not set by default.
+;waitfordialtone=1000
 ;
 ; The following option enables receiving MWI on FXO lines.  The default
 ; value is no.
@@ -931,6 +944,33 @@
 ; buffer policy.
 ;
 ;faxbuffers=>6,full
+;
+; Configure the default number of DAHDI buffers and the transmit policy to use.
+; This can be used to eliminate data drops when scheduling jitter prevents
+; Asterisk from writing to a DAHDI channel regularly. Most users will probably
+; want "faxbuffers" instead of "buffers".
+;
+; The policies are:
+; immediate - DAHDI will immediately start sending the data to the hardware after
+;             Asterisk writes to the channel. This is the default mode. It
+;             introduces the least amount of latency but has an increased chance for
+;             hardware under runs if Asterisk is not able to keep the DAHDI write
+;             queue from going empty.
+; half      - DAHDI will wait until half of the configured buffers are full before
+;             starting to transmit. This adds latency to the audio but reduces
+;             the chance of under runs. Essentially, this is like an in-kernel jitter
+;             buffer.
+; full      - DAHDI will not start transmitting until all buffers are full.
+;             Introduces the most amount of latency and is susceptible to over
+;             runs from the Asterisk process.
+;
+; The receive policy is never changed. DAHDI will always pass up audio as soon
+; as possible.
+;
+; The default number of buffers is 4 (from jitterbuffers) and the default policy
+; is immediate.
+;
+;buffers=4,immediate
 ;
 ; This option specifies a preference for which music on hold class this channel
 ; should listen to when put on hold if the music class has not been set on the

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/dsp.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/dsp.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/dsp.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/dsp.conf.sample Tue Jun  3 11:06:55 2014
@@ -5,3 +5,39 @@
 ; to talking.  [default=256]
 ;
 ;silencethreshold=256
+
+; DTMF Reverse Twist and Normal Twist is the difference in power between the row and column energies.
+;
+; Normal Twist is where the Column energy is greater than the Row energy
+; Reverse Twist is where the Row energy is greater.
+;
+; Power level difference between frequencies for different Administrations/RPOAs
+;		Power Gain		equiv
+;		normal	reverse		dB's
+; AT&T(default) 6.31	2.51		8dB(normal), 4dB(reverse)
+; NTT		3.16	3.16		Max. 5dB
+; Danish	3.98	3.98		Max. 6dB
+; Australian	10.0	10.0		Max. 10dB
+; Brazilian	7.94	7.94		Max. 9dB
+; ETSI		3.98	3.98		Max. 6dB
+
+;previous version compatible AT&T values
+; RADIO_RELAX disabled, and relaxdtmf=no
+;		6.30	2.50		7.99dB(normal), 3.98dB(reverse)
+; RADIO_RELAX disabled, and relaxdtmf=yes
+;		6.30	4.00		7.99dB(normal), 6.02dB(reverse)
+; RADIO_RELAX enabled, and relaxdtmf=no
+;		6.30	2.50		7.99dB(normal), 3.984dB(reverse)
+; RADIO_RELAX enabled, and relaxdtmf=yes
+;		6.30	6.50		7.99dB(normal), 8.13dB(reverse)
+
+;If you don't know what these mean, don't change them.
+;dtmf_normal_twist=6.31
+;dtmf_reverse_twist=2.51
+;relax_dtmf_normal_twist=6.31
+;relax_dtmf_reverse_twist=3.98
+
+;successive number hits/misses of 12.75ms before a digit/nodigit is considered valid
+;dtmf_hits_to_begin=2
+;dtmf_misses_to_end=3
+

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/extconfig.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/extconfig.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/extconfig.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/extconfig.conf.sample Tue Jun  3 11:06:55 2014
@@ -53,13 +53,17 @@
 ; start at 1 and be sequential (i.e. if you have only priorities 1, 2,
 ; and 4, then 4 will be ignored, because there is no 3).
 ;
+;
+; Possible driver backends:
+;
 ; "odbc" is shown in the examples below, but is not the only valid realtime
-; engine.  There is:
+; engine.  Here are several of the possible options:
 ;    odbc ... res_config_odbc
 ;    sqlite ... res_config_sqlite
 ;    pgsql ... res_config_pgsql
 ;    curl ... res_config_curl
 ;    ldap ... res_config_ldap
+;    mysql ... res_config_mysql (available via add-ons in menuselect)
 ;
 ; Note: The res_config_pgsql and res_config_sqlite backends configure the
 ; database used in their respective configuration files and ignore the

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/h323.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/h323.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/h323.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/h323.conf.sample Tue Jun  3 11:06:55 2014
@@ -29,6 +29,8 @@
 ;allow=gsm		; Always allow GSM, it's cool :)
 ;allow=ulaw		; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
 			; for framing options
+;autoframing=yes	; Set packetization based on the remote endpoint's (ptime)
+			; preferences. Defaults to no.
 ;
 ; User-Input Mode (DTMF)
 ;

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/iax.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/iax.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/iax.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/iax.conf.sample Tue Jun  3 11:06:55 2014
@@ -345,12 +345,12 @@
 ;
 ; Call token validation can be set as optional for a single IP address or IP
 ; address range by using the 'calltokenoptional' option. 'calltokenoptional' is
-; only a global option.  
+; only a global option.
 ;
 ;calltokenoptional=209.16.236.73/255.255.255.0
 ;
 ; By setting 'requirecalltoken=no', call token validation becomes optional for
-; that peer/user.  By setting 'requirecalltoken=auto', call token validation 
+; that peer/user.  By setting 'requirecalltoken=auto', call token validation
 ; is optional until a call token supporting peer registers successfully using
 ; call token validation.  This is used as an indication that from now on, we
 ; can require it from this peer.  So, requirecalltoken is internally set to yes.
@@ -380,7 +380,7 @@
 ; has been disabled.  Unlike the 'maxcallnumbers' option, this limit is not
 ; separate for each individual IP address.  Any connection resulting in a
 ; non-call token validated call number being allocated contributes to this
-; limit.  For use cases, see the call token user guide.  This option's 
+; limit.  For use cases, see the call token user guide.  This option's
 ; default value of 8192 should be sufficient in most cases.
 ;
 ;maxcallnumbers_nonvalidated=1024
@@ -389,7 +389,7 @@
 ; for specific IP addresses and IP address ranges.  These limits take precedence
 ; over the global 'maxcallnumbers' option, but may still be overridden by a
 ; peer defined 'maxcallnumbers' entry.  Note that these limits take effect
-; for every individual address within the range, not the range as a whole. 
+; for every individual address within the range, not the range as a whole.
 ;
 ;[callnumberlimits]
 ;10.1.1.0/255.255.255.0 = 24
@@ -530,6 +530,19 @@
 ; suggested to the other side as well if it is for example a phone instead of
 ; another PBX.
 ;
+;connectedline=yes ; Set if connected line and redirecting information updates
+;                  ; are passed between Asterisk servers for this peer.
+;                  ; yes - Sending and receiving updates are enabled.
+;                  ; send - Only send updates.
+;                  ; receive - Only process received updates.
+;                  ; no - Sending and receiving updates are disabled.
+;                  ; Default is "no".
+;                  ;
+;                  ; Note: Because of an incompatibility between Asterisk v1.4
+;                  ; and Asterisk v1.8 or later, this option must be set
+;                  ; to "no" toward the Asterisk v1.4 peer.  A symptom of the
+;                  ; incompatibility is the call gets disconnected unexpectedly.
+
 
 ;[dynamichost]
 ;host=dynamic

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/indications.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/indications.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/indications.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/indications.conf.sample Tue Jun  3 11:06:55 2014
@@ -285,6 +285,8 @@
 record = 1400/500,0/15000
 info = 950/330,0/1000
 dialout = 500
+; STUTTER not specified
+stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
 
 
 [fi]
@@ -371,7 +373,7 @@
 dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
 record = 1400/500,0/15000
 info = !950/330,!1400/330,!1800/330,0/1000
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,400*25
 
 [it]
 description = Italy
@@ -427,15 +429,17 @@
 dialrecall = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
 record = 1400/500,0/15000
 info = 950/330,0/30,1400/330,0/30,1800/330,0/1000
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,425
 
 [my]
 description = Malaysia
 ringcadence = 2000,4000
 dial = 425
 busy = 425/500,0/500
-ring = 425/400,0/200
+ring = 425/400,0/200,425/400,0/2000
 congestion = 425/500,0/500
+; STUTTER - not specified
+stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
 
 [nl]
 description = Netherlands
@@ -500,7 +504,7 @@
 ; INFO - not specified
 info = !950/330,!1400/330,!1800/330,0
 ; STUTTER - not specified
-stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,350+440
+stutter = !350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,!350+440/100,!0/100,425
 
 
 [pl]
@@ -702,6 +706,8 @@
 dialrecall = 425
 record = 1400/500,0/15000
 info = !950/330,!1440/330,!1800/330,0/1000
+; STUTTER - not specified
+stutter = !425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,!425/100,!0/100,425
 
 
 [za]

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/manager.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/manager.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/manager.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/manager.conf.sample Tue Jun  3 11:06:55 2014
@@ -87,12 +87,13 @@
 ;permit=209.16.236.73/255.255.255.0
 ;
 ;eventfilter=Event: Newchannel
-;eventfilter=!Channel: DAHDI*
-; The eventfilter option is used to whitelist or blacklist events per user to be
-; reported with regular expressions and are allowed if both the regex matches
-; and the user has read access set below. Filters are assumed to be for whitelisting
-; unless preceeded by an exclamation point, which marks it as being black.
-; Evaluation of the filters is as follows:
+;eventfilter=!Channel: DAHDI.*
+; The eventfilter option is used to whitelist or blacklist events per user.
+; A filter consists of a (basic/old-style and unanchored) regular expression
+; that is run on the entire event data. If the first character of the filter
+; is an exclamation mark (!), the filter is appended to the blacklist instead
+; of the whitelist. After first checking the read access below, the regular
+; expression filters are processed as follows:
 ; - If no filters are configured all events are reported as normal.
 ; - If there are white filters only: implied black all filter processed first,
 ; then white filters.

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/queues.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/queues.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/queues.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/queues.conf.sample Tue Jun  3 11:06:55 2014
@@ -5,7 +5,7 @@
 ; Persistent Members
 ;    Store each dynamic member in each queue in the astdb so that
 ;    when asterisk is restarted, each member will be automatically
-;    read into their recorded queues. Default is 'yes'.
+;    read into their recorded queues. Default is 'no'.
 ;
 persistentmembers = yes
 ;
@@ -57,7 +57,7 @@
 ; shared_lastcall will make the lastcall and calls received be the same in
 ; members logged in more than one queue.  This is useful to make the queue
 ; respect the wrapuptime of another queue for a shared member.
-; The default value is yes.
+; The default value is no.
 ;
 ;shared_lastcall=no
 ;
@@ -292,6 +292,13 @@
 ; will have their position announced.
 ;
 ;announce-position = yes
+;
+; If enabled, play announcements to the first user waiting in the Queue. This may mean
+; that announcements are played when an agent attempts to connect to the waiting user,
+; which may delay the time before the agent and the user can communicate. Disabled by
+; default.
+;
+; announce-to-first-user = no
 ;
 ; If you have specified "limit" or "more" for the announce-position option, then the following
 ; value is what is used to determine what announcement to play to waiting callers. If you have

Propchange: team/oej/adb-appleraision-1.8-mark-2/configs/res_curl.conf.sample
------------------------------------------------------------------------------
--- svn:keywords (original)
+++ svn:keywords Tue Jun  3 11:06:55 2014
@@ -1,1 +1,1 @@
-'Date Author Id Revision Yoyo'
+Author Date Id Revision

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/res_fax.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/res_fax.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/res_fax.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/res_fax.conf.sample Tue Jun  3 11:06:55 2014
@@ -4,12 +4,12 @@
 ; Maximum Transmission Rate
 ; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
 ; Set this value to the maximum desired transfer rate.  Default: 14400
-maxrate=14400
+;maxrate=14400
 
 ; Minimum Transmission Rate
 ; Possible values are { 2400 | 4800 | 7200 | 9600 | 12000 | 14400 }
-; Set this value to the minimum desired transfer rate.  Default: 2400
-minrate=2400
+; Set this value to the minimum desired transfer rate.  Default: 4800
+;minrate=4800
 
 ; Send Progress/Status events to manager session
 ; Manager events with 'call' class permissions will receive events indicating the
@@ -21,8 +21,8 @@
 ; modem capabilities
 ; Possible values are { v17 | v27 | v29 }
 ; Set this value to modify the default modem options.  Default: v17,v27,v29
-modems=v17,v27,v29
+;modems=v17,v27,v29
 
 ; Enable/disable T.30 ECM (error correction mode) by default.
 ; Default: Enabled
-ecm=yes
+;ecm=yes

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/res_ldap.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/res_ldap.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/res_ldap.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/res_ldap.conf.sample Tue Jun  3 11:06:55 2014
@@ -13,6 +13,8 @@
 ;
 ; In the case of LDAP the last keyword in each line above specifies
 ; a section in this file.
+;
+; LDAP schema and ldif files can be located in contrib/scripts.
 
 ; TLS support
 ; -----------

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/res_odbc.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/res_odbc.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/res_odbc.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/res_odbc.conf.sample Tue Jun  3 11:06:55 2014
@@ -23,7 +23,8 @@
 ; (or /usr/local/etc/odbc.ini, on FreeBSD and similar systems).
 dsn => asterisk
 ;
-; Username for connecting to the database.  The default user is "root".
+; Username for connecting to the database.  The user defaults to the context
+; name if unspecified.
 ;username => myuser
 ;
 ; Password for authenticating the user to the database.  The default
@@ -63,6 +64,15 @@
 ; Is the backslash a native escape character?  The default is yes, but for
 ; MS SQL Server, the answer is no.
 ;backslash_is_escape => yes
+;
+; When enabled (default behavior), empty column values are stored as empty strings
+; during realtime updates. Disabling this option causes empty column values to be
+; stored as NULLs for non-text columns.
+; Disable it for PostgreSQL backend in order to avoid errors caused by updating
+; integer columns with an empty string instead of NULL (sippeers, sipregs, ..).
+; NOTE: This option will be removed in asterisk 13. At that point, it will always
+; behave as if it was set to 'no'.
+;allow_empty_string_in_nontext => yes
 ;
 ; How long (in seconds) should we attempt to connect before considering the
 ; connection dead?  The default is 10 seconds, but you may wish to reduce it,

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/sip.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/sip.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/sip.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/sip.conf.sample Tue Jun  3 11:06:55 2014
@@ -300,6 +300,8 @@
 ;allow=ulaw                     ; Allow codecs in order of preference
 ;allow=ilbc                     ; see https://wiki.asterisk.org/wiki/display/AST/RTP+Packetization
 				; for framing options
+;autoframing=yes		; Set packetization based on the remote endpoint's (ptime)
+				; preferences. Defaults to no.
 ;
 ; This option specifies a preference for which music on hold class this channel
 ; should listen to when put on hold if the music class has not been set on the
@@ -335,6 +337,17 @@
                                 ; transmit such UPDATE messages to it, then you must enable this option.
                                 ; Otherwise, we will have to wait until we can send a reinvite to
                                 ; transmit the information.
+;trust_id_outbound = no         ; Controls whether or not we trust this peer with private identity
+                                ; information (when the remote party has callingpres=prohib or equivalent).
+                                ; no - RPID/PAI headers will not be included for private peer information
+                                ; yes - RPID/PAI headers will include the private peer information. Privacy
+                                ;       requirements will be indicated in a Privacy header for sendrpid=pai
+                                ; legacy - RPID/PAI will be included for private peer information. In the
+                                ;       case of sendrpid=pai, private data that would be included in them
+                                ;       will be anonymized. For sendrpid=rpid, private data may be included
+                                ;       but the remote party's domain will be anonymized. The way legacy
+                                ;       behaves may violate RFC-3325, but it follows historic behavior.
+                                ; This option is set to 'legacy' by default
 ;prematuremedia=no              ; Some ISDN links send empty media frames before 
                                 ; the call is in ringing or progress state. The SIP 
                                 ; channel will then send 183 indicating early media
@@ -381,6 +394,9 @@
                                 ; certain transferred calls to use always use video when
                                 ; available. [yes|NO|always]
 
+;textsupport=no                 ; Support for ITU-T T.140 realtime text.
+                                ; The default value is "no".
+
 ;maxcallbitrate=384             ; Maximum bitrate for video calls (default 384 kb/s)
                                 ; Videosupport and maxcallbitrate is settable
                                 ; for peers and users as well
@@ -428,7 +444,7 @@
 ;contactpermit=172.16.0.0/255.255.0.0  ; restrict at what IPs your users may
                                        ; register their phones.
 
-;engine=asterisk                ; RTP engine to use when communicating with the device
+;rtp_engine=asterisk            ; RTP engine to use when communicating with the device
 
 ;
 ; If regcontext is specified, Asterisk will dynamically create and destroy a
@@ -469,8 +485,10 @@
                       ; Set to yes add Reason header and use Reason header if it is available.
 ;
 ;------------------------ TLS settings ------------------------------------------------------------
-;tlscertfile=</path/to/certificate.pem> ; Certificate file (*.pem format only) to use for TLS connections
-                                        ; default is to look for "asterisk.pem" in current directory
+;tlscertfile=</path/to/certificate.pem> ; Certificate chain (*.pem format only) to use for TLS connections
+                                        ; The certificates must be sorted starting with the subject's certificate
+                                        ; and followed by intermediate CA certificates if applicable.
+                                        ; Default is to look for "asterisk.pem" in current directory
 
 ;tlsprivatekey=</path/to/private.pem> ; Private key file (*.pem format only) for TLS connections.
                                       ; If no tlsprivatekey is specified, tlscertfile is searched for
@@ -545,11 +563,20 @@
 ; * session-expires   - Maximum session refresh interval in seconds. Defaults to 1800 secs.
 ; * session-minse     - Minimum session refresh interval in seconds. Defualts to 90 secs.
 ; * session-refresher - The session refresher (uac|uas). Defaults to 'uas'.
+;                            uac - Default to the caller initially refreshing when possible
+;                            uas - Default to the callee initially refreshing when possible
+;
+; Note that, due to recommendations in RFC 4028, Asterisk will always honor the other
+; endpoint's preference for who will handle refreshes. Asterisk will never override the
+; preferences of the other endpoint. Doing so could result in Asterisk and the endpoint
+; fighting over who sends the refreshes. This holds true for the initiation of session
+; timers and subsequent re-INVITE requests whether Asterisk is the caller or callee, or
+; whether Asterisk is currently the refresher or not.
 ;
 ;session-timers=originate
 ;session-expires=600
 ;session-minse=90
-;session-refresher=uas
+;session-refresher=uac
 ;
 ;--------------------------- SIP DEBUGGING ---------------------------------------------------
 ;sipdebug = yes                 ; Turn on SIP debugging by default, from
@@ -711,6 +738,9 @@
                                 ; 0 = continue forever, hammering the other server
                                 ; until it accepts the registration
                                 ; Default is 0 tries, continue forever
+;register_retry_403=yes         ; Treat 403 responses to registrations as if they were
+                                ; 401 responses and continue retrying according to normal
+                                ; retry rules.
 
 ;----------------------------------------- OUTBOUND MWI SUBSCRIPTIONS -------------------------
 ; Asterisk can subscribe to receive the MWI from another SIP server and store it locally for retrieval
@@ -876,6 +906,13 @@
 ;directmedia=update             ; Yet a third option... use UPDATE for media path redirection,
                                 ; instead of INVITE. This can be combined with 'nonat', as
                                 ; 'directmedia=update,nonat'. It implies 'yes'.
+
+;directmedia=outgoing           ; When sending directmedia reinvites, do not send an immediate
+                                ; reinvite on an incoming call leg. This option is useful when
+                                ; peered with another SIP user agent that is known to send
+                                ; immediate direct media reinvites upon call establishment. Setting
+                                ; the option in this situation helps to prevent potential glares.
+                                ; Setting this option implies 'yes'.
 
 ;directrtpsetup=yes             ; Enable the new experimental direct RTP setup. This sets up
                                 ; the call directly with media peer-2-peer without re-invites.
@@ -1105,8 +1142,10 @@
 ; language
 ; allow
 ; disallow
+; autoframing
 ; insecure
 ; trustrpid
+; trust_id_outbound
 ; progressinband
 ; promiscredir
 ; useclientcode
@@ -1145,7 +1184,6 @@
 ; outboundproxy
 ; rfc2833compensate
 ; callbackextension
-; registertrying
 ; timert1
 ; timerb
 ; qualifyfreq
@@ -1275,7 +1313,8 @@
 ;allow=g723.1                    ; Asterisk only supports g723.1 pass-thru!
 ;allow=g729                      ; Pass-thru only unless g729 license obtained
 ;callingpres=allowed_passed_screen ; Set caller ID presentation
-                                 ; See README.callingpres for more information
+                                 ; See function CALLERPRES documentation for possible
+                                 ; values.
 
 ;[xlite1]
 ; Turn off silence suppression in X-Lite ("Transmit Silence"=YES)!

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/sip_notify.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/sip_notify.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/sip_notify.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/sip_notify.conf.sample Tue Jun  3 11:06:55 2014
@@ -16,6 +16,11 @@
 
 [aastra-xml]
 Event=>aastra-xml
+
+; Digium
+
+[digium-check-cfg]
+Event=>check-sync
 
 ; Linksys
 

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/sla.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/sla.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/sla.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/sla.conf.sample Tue Jun  3 11:06:55 2014
@@ -13,6 +13,17 @@
                             ; feel free to enable it if you would like.  If you do, and
                             ; you find problems, please do not report them.
 ; -------------------------------------
+
+
+; ********************************
+; **** Configuration Ordering ****
+; ********************************
+
+; Note that SLA configuration processing assumes that *all* trunk declarations are
+; listed in the configuration file before any stations.
+
+; ********************************
+; ********************************
 
 
 ; ---- Trunk Declarations -------------

Modified: team/oej/adb-appleraision-1.8-mark-2/configs/voicemail.conf.sample
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/configs/voicemail.conf.sample?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/configs/voicemail.conf.sample (original)
+++ team/oej/adb-appleraision-1.8-mark-2/configs/voicemail.conf.sample Tue Jun  3 11:06:55 2014
@@ -169,7 +169,8 @@
 ; Short 24h date format for pager use
 ;pagerdateformat=%T %D
 ;
-; You can override the default program to send e-mail if you wish, too
+; Using the mailcmd option, you can specify what command is called for
+; outbound E-mail. The default is shown below.
 ;
 ;mailcmd=/usr/sbin/sendmail -t
 ;

Modified: team/oej/adb-appleraision-1.8-mark-2/funcs/func_odbc.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/adb-appleraision-1.8-mark-2/funcs/func_odbc.c?view=diff&rev=415057&r1=415056&r2=415057
==============================================================================
--- team/oej/adb-appleraision-1.8-mark-2/funcs/func_odbc.c (original)

[... 7222 lines stripped ...]



More information about the asterisk-commits mailing list